Audio compressors are well known devices that may be advantageously employed in devices for assisting the hearing impaired. In a compressor, the gain of the device varies as a function of the amplitude of the input signal, the gain being greatest for a low level input signal and smallest for a large amplitude signal.
In many cases the response of the ear of a hearing impaired person will be substantially different in terms of sensitivity and frequency response from that of a normal person. For a person with what is termed sensorineural hearing impairment, soft sounds are rendered inaudible, while loud sounds may be subjectively just as loud as for a normal person. Conversational levels may be very soft or even inaudible for a person with sensorineural hearing impairment. Consequently, if linear amplification is used to assist such a person, loudness relationships are perceived as distorted, and loud sounds may be rendered uncomfortably and, in some cases, painfully loud. What is necessary for comprehension of speech by the hearing impaired is to raise the amplitude of soft speech cues to the level of audibility. Beyond this, further improvements may be had by reestablishing loudness relationships. These concepts may be further understood by referring to Edgar Villchur, "Signal processing to improve speech intelligibility in perceptive deafness," Journal of the Acoustical Society of America, Vol. 53, pp. 1646-1657.
Studies have shown that hearing aids with audio compression provide improved syllabic comprehension for persons with sensorineural hearing losses. The use of audio compressors for the hearing impaired is described extensively in a report written by Walker and Dillon, entitled "Compression in Hearing Aids: An Analysis, A Review, and Some Recommendations", NAL Report No. 90, published by the Australian Commonwealth Department of Health, National Acoustics Laboratories, June 1982. Another clinical study, recently completed by B. C. J. Moore, J. S. Johnson, T. C. Clark, and V. Pluvinage is reported in a paper, "Evaluation of a dual channel full dynamic range system for people with sensorineural hearing loss," to be published in the Journal of the Acoustical Society of America. This paper is conclusive on the benefits of audio multiband compression.
Audio compressors may also be advantageously employed to tailor the characteristics of a hearing aid device to compensate for unique deficiencies of individual users as well as to simulate normal hearing under a variety of situations such as quiet or noisy environments. For example, in many cases a hearing impaired person will only experience a hearing loss at high frequencies and at low levels. For such a person it is desirable to provide a device which amplifies sound only at low levels and high frequencies. At high levels and high frequencies, the gain of the device is reduced typically to a value close to unity. At low frequencies, the gain may be held near unity for all input sound levels. Thus, for this case, compression is introduced at high frequencies only, and the gain is near unity at high sound levels throughout the audio spectrum.
Since each hearing impaired person has a unique hearing response, a compression system with adjustable compression ratio and frequency response is highly desirable. A multiband compressor system of wide dynamic range is described in U.S. Pat. No. 4,882,762 titled "Multi-Band Programmable Compression System", issued Nov. 21, 1989, to the present applicant. This patent (the disclosure of which is hereby incorporated by reference), describes a system, currently in manufacture, for improving the hearing of patients with sensorineural hearing impairment. The system consists of an input transducer to convert ambient acoustic signals to electrical signals, electronic amplifier stages to establish appropriate signal levels at various points in the system, a multiband compressor, and an output transducer to convert the amplified electrical signals back into acoustic or mechanical form which can be heard by the hearing impaired person.
The multiband compressor uses a plurality of compressor circuits of the type described in U.S. Pat. No. 4,882,761, "A Low Voltage Programmable Compressor," issued Nov. 28, 1989, also to present applicant. The compressor circuits shown in the '761 patent, (the disclosure of which is also hereby incorporated by reference), each perform effectively three functions in their respective frequency bands: first, each circuit acts as a compressor circuit with programmable compression ratio; second, each provides an amount of gain that is programmable to allow tailoring to the individual user; and third, each provides the user volume control function. For purposes of this discussion, the term compression ratio is the ratio of a change in input signal level to the resulting change in output level.
The wide range of gains that must be provided by the compressor circuits in performing these three functions has been found to adversely affect their performance, particularly as compressors, as described below.
In view of the extremely small size required of hearing device circuits which must fit in the human ear, the design concept of the device shown in the '761 patent was to incorporate these multiple functions in a single circuit. The improved compressor circuit described herein removes this limitation by separating these functions into separate circuits, thereby greatly improving performance quality, reducing distortion, increasing flexibility of gain, simplifying programmability and increasing manufacturing yields, while taking no more silicon area on an integrated circuit "chip" than the earlier design.
To appreciate how the circuit performs its function, we first describe the workings of a limiter having programmable characteristics as required in the present invention and how such a limiter is used in a compressor having a programmable compression ratio.
In order to provide signal limiting, a variable gain circuit controllable by an electrical signal is required. Such circuits are well known in the art; one circuit that is particularly adaptable to the low supply voltages used in hearing aids is described in U.S. Pat. No. 4,868,517, Variolosser, issued Sep. 19, 1989 to the present applicant and Baez. As described in the '517 patent (the disclosure of which is hereby incorporated by reference), the current gain of the variolosser is equal to the ratio of two control currents, designated as I.sub.A and I.sub.B, with the overall current gain equal to I.sub.B /I.sub.A. If we rectify and filter the incoming signal to this circuit to provide an envelope signal, and use this envelope signal to increase I.sub.A as the input signal increases, the current gain will decrease as the input signal increases. Hence, the slope of a curve of the output level vs input level begins to diminish at higher levels, until the output level tends to remain constant above a given input level.
The gain falls as the input level increases because I.sub.A, in the ratio I.sub.B /I.sub.A, increases with input level. I.sub.A has two components, a fixed dc current and a signal dependent current. At very low sound levels, the fixed dc component of I.sub.A predominates so that the gain is constant there. It is only when the signal dependent part of I.sub.A becomes significant in relation to the dc part that the gain begins to fall. We define P.sub.lb, the lower break point, as the sound level at which the signal dependent part of I.sub.A becomes equal to the dc part. Note that so far we have described what may be referred to as a "soft limiter", i.e. , a limiter in which the gain changes gradually as the input signal increases. In this soft limiter I.sub.B is a fixed or DC current. From our definition of the lower break point, it can be seen that the response at the break point is 6 dB below the intersection of the asymptotic values of the curve as the input level goes to zero. This follows from the fact that at the break point I.sub.A0 =I.sub.As and I.sub.A =I.sub.A0 +I.sub.As, where I.sub.A0 is the dc or constant component of I.sub.A, and I.sub.As is the signal related component of I.sub.A. Thus, at the break point I.sub.A =2I.sub.A0, i.e. , the overall gain is reduced by 50% or 6 dB.
In the programmable compressor of the above mentioned '761 patent, the limiting action causes the slope of the output level vs input level at higher levels to begin to diminish as the value of I.sub.A rises above its dc value as described above. However, in the '761 device at some higher input level the value of I.sub.B is caused to rise above its dc value under the influence of the input signal. This introduces a second, upward break point into the curve that we term P.sub.ub, the upper break point. At higher levels beyond P.sub.ub the slope of the output level vs input level begins to asymptotically approach a linear value. The mathematical function that defines these characteristics, i.e. , a sigmoid function, tends to be almost linear in the region between the lower and upper break points, with a point of inflection midway between these break points, as described below. The input level corresponding to this inflection point will be termed P.sub.V, the pivot point. By varying the distance between the upper and lower break points, P.sub.lb and P.sub.ub, we can vary the slope of the output-input curve between the break points over a relatively wide range of audio input amplitude. It can be shown that the reciprocal of the slope of the input-output curve is the compression ratio, .mu.. Thus, by varying P.sub.ub relative to P.sub.lb (or vice versa), it is possible to obtain a continuous variation of the compression ratio. Programming the compression ratio can therefore be accomplished by variation of one or both of these break points, which, it will be remembered, are set by dc currents applied to the aforementioned variolosser.
The prior art variable compressor applied the rectified envelope signal to change both I.sub.A and I.sub.B, whereas in accordance with the present invention, I.sub.A provides the limiting action, while I.sub.B provides for programming the circuit to provide a desired compression ratio. This improvement has several other advantages that will be discussed below.
In addition to providing programmable compression ratios, the prior art circuit also allowed programming of gain, simply by varying the fixed or dc portions of I.sub.A and I.sub.B, i.e., lowering I.sub.A0 or increasing I.sub.B0 causes the gain to increase.
Because of the wide range over which the gain of these circuits must vary in the prior art system in providing the three aforementioned functions, the values of control currents I.sub.A and I.sub.B must vary over a fairly wide range. This was found to lead to intermodulation between the control signals and spurious dc offset voltages leading to distortions, discussed below, that became quite severe for some gain or compression programmed values when the circuits were not exactly in balance. Thus, while the circuit of the '761 patent works very well to provide the desired functions, it has been found that extremely careful adjustment of the circuit parameters is required to maintain high quality signal processing. By requiring that I.sub.A and I.sub.B vary over a range wide enough to fulfill the demands of different amounts of hearing impairment, exacting specifications are placed on the integrated circuit processing to obtain high quality sound. These exacting specifications increase the manufacturing expense and reduce the manufacturing yield.